Added logging for audio send/receive stream configs.

webrtc/tools/event_log_visualizer/analyzer.cc

 /*
  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 335 matching lines @@
           StreamId rtx_stream(ssrc, kOutgoingPacket);
           RegisterHeaderExtensions(config.rtp.extensions,
                                    &extension_maps[rtx_stream]);
           video_ssrcs_.insert(rtx_stream);
           rtx_ssrcs_.insert(rtx_stream);
         }
         break;
       }
       case ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT: {
         AudioReceiveStream::Config config;
-        // TODO(terelius): Parse the audio configs once we have them.
+        parsed_log_.GetAudioReceiveConfig(i, &config);
+        StreamId stream(config.rtp.remote_ssrc, kIncomingPacket);
+        RegisterHeaderExtensions(config.rtp.extensions,
+                                 &extension_maps[stream]);
+        audio_ssrcs_.insert(stream);
         break;
       }
       case ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT: {
         AudioSendStream::Config config(nullptr);
-        // TODO(terelius): Parse the audio configs once we have them.
+        parsed_log_.GetAudioSendConfig(i, &config);
+        StreamId stream(config.rtp.ssrc, kOutgoingPacket);
+        RegisterHeaderExtensions(config.rtp.extensions,
+                                 &extension_maps[stream]);
+        audio_ssrcs_.insert(stream);
         break;
       }
       case ParsedRtcEventLog::RTP_EVENT: {
         MediaType media_type;
         parsed_log_.GetRtpHeader(i, &direction, &media_type, header,
                                  &header_length, &total_length);
         // Parse header to get SSRC.
         RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
         RTPHeader parsed_header;
         rtp_parser.Parse(&parsed_header);
@@ skipping 787 matching lines @@
     point.y -= estimated_base_delay_ms;
   // Add the data set to the plot.
   plot->series_list_.push_back(std::move(time_series));
 
   plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
   plot->SetSuggestedYAxis(0, 10, "Delay (ms)", kBottomMargin, kTopMargin);
   plot->SetTitle("Network Delay Change.");
 }
 }  // namespace plotting
 }  // namespace webrtc